applied final release of bug 1353 per Mark's permission

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2782 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 106 insertions, 3 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index af3c70c6d..3492c1004 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -20,6 +20,7 @@
 #include <asterisk/module.h>
 #include <asterisk/translate.h>
 #include <asterisk/say.h>
+#include <asterisk/config.h>
 #include <asterisk/parking.h>
 #include <asterisk/musiconhold.h>
 #include <asterisk/callerid.h>
@@ -68,6 +69,15 @@ static char *descrip =
 "      'g' -- goes on in context if the destination channel hangs up\n"
 "      'A(x)' -- play an announcement to the called party, using x as file\n"
 "      'S(x)' -- hangup the call after x seconds AFTER called party picked up\n"  	
+"      'L(x[:y][:z])' -- Limit the call to 'x' ms warning when 'y' ms are left (repeated every 'z' ms)\n"
+"                     -- Only 'x' is required, 'y' and 'z' are optional.\n"
+"                     -- The following special variables are optional:\n"
+"                       ** LIMIT_PLAYAUDIO_CALLER    (default yes) Play sounds to the caller.\n"
+"                       ** LIMIT_PLAYAUDIO_CALLEE    Play sounds to the callee.\n"
+"                       ** LIMIT_TIMEOUT_FILE        File to play when time is up.\n"
+"                       ** LIMIT_CONNECT_FILE        File to play when call begins.\n"
+"                       ** LIMIT_WARNING_FILE        File to play as warning if 'y' is defined.\n"
+"                     -- 'timeleft' is a special sound macro to auto-say the time left and is the default.\n\n"
 "  In addition to transferring the call, a call may be parked and then picked\n"
 "up by another user.\n"
 "  The optional URL will be sent to the called party if the channel supports\n"
@@ -390,7 +400,19 @@ static int dial_exec(struct ast_channel *chan, void *data)
 	unsigned int calldurationlimit=0;
 	char *cdl;
 	time_t now;
-	
+	struct ast_bridge_config config;
+	long timelimit = 0;
+	long play_warning = 0;
+	long warning_freq=0;
+	char *warning_sound=NULL;
+	char *end_sound=NULL;
+	char *start_sound=NULL;
+	char *limitptr;
+	char limitdata[256];
+	char *stack,*var;
+	int play_to_caller=0,play_to_callee=0;
+	int playargs=0;
+
 	if (!data) {
 		ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout|options)\n");
 		return -1;
@@ -430,13 +452,81 @@ static int dial_exec(struct ast_channel *chan, void *data)
 	
 
 	if (transfer) {
+
 		/* Extract call duration limit */
 		if ((cdl = strstr(transfer, "S("))) {
 			calldurationlimit=atoi(cdl+2);
 			if (option_verbose > 2)
 				ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %i seconds.\n",calldurationlimit);			
 		} 
-		
+
+		/* XXX LIMIT SUPPORT */
+		if ((limitptr = strstr(transfer, "L("))) {
+			strncpy(limitdata, limitptr + 2, sizeof(limitdata) - 1);
+			/* Overwrite with X's what was the limit info */
+			while(*limitptr && (*limitptr != ')')) 
+				*(limitptr++) = 'X';
+            if (*limitptr)
+                *limitptr = 'X';
+            /* Now find the end of the privdb */
+            limitptr = strchr(limitdata, ')');
+            if (limitptr)
+                *limitptr = '\0';
+            else {
+                ast_log(LOG_WARNING, "Limit Data lacking trailing ')'\n");
+            }
+
+            var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER");
+            play_to_caller = var ? ast_true(var) : 1;
+
+            var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLEE");
+            play_to_callee = var ? ast_true(var) : 0;
+            
+            if(! play_to_caller && ! play_to_callee)
+                play_to_caller=1;
+
+            var = pbx_builtin_getvar_helper(chan,"LIMIT_WARNING_FILE");
+            warning_sound = var ? var : "timeleft";
+            
+            var = pbx_builtin_getvar_helper(chan,"LIMIT_TIMEOUT_FILE");
+            end_sound = var ? var : NULL;
+
+            var = pbx_builtin_getvar_helper(chan,"LIMIT_CONNECT_FILE");
+            start_sound = var ? var : NULL;
+            
+            var=stack=limitdata;
+            
+            var = strsep(&stack, ":");
+            if(var) {
+                timelimit = atol(var);
+                playargs++;
+            }
+            var = strsep(&stack, ":");
+            if(var) {
+                play_warning = atol(var);
+                playargs++;
+            }
+
+            var = strsep(&stack, ":");
+            if(var) {
+                warning_freq = atol(var);
+                playargs++;
+            }
+            
+            if(! timelimit) {
+                timelimit=play_to_caller=play_to_callee=play_warning=warning_freq=0;
+                warning_sound=NULL;
+            }
+            calldurationlimit=0; /* undo effect of S(x) in case they are both used */
+            if(! play_warning && ! start_sound && ! end_sound && timelimit) { /* more efficient do it like S(x) does since no advanced opts*/
+				calldurationlimit=timelimit/1000;
+				timelimit=play_to_caller=play_to_callee=play_warning=warning_freq=0;
+			}
+			else 
+                ast_verbose(VERBOSE_PREFIX_3"Limit Data: timelimit=%ld\n    -- play_warning=%ld\n    -- play_to_caller=%s\n    -- play_to_callee=%s\n    -- warning_freq=%ld\n    -- warning_sound=%s\n    -- end_sound=%s\n    -- start_sound=%s\n",timelimit,play_warning,play_to_caller ? "yes" : "no",play_to_callee ? "yes" : "no",warning_freq,warning_sound ? warning_sound : "UNDEF",end_sound ? end_sound : "UNDEF",start_sound ? start_sound : "UNDEF");
+				
+		}
+
 		/* XXX ANNOUNCE SUPPORT */
 		if ((ann = strstr(transfer, "A("))) {
 			announce = 1;
@@ -729,7 +819,20 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			time(&now);
 			chan->whentohangup = now + calldurationlimit;
 		}
-		res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect);
+
+		memset(&config,0,sizeof(struct ast_bridge_config));
+		config.play_to_caller=play_to_caller;
+		config.play_to_callee=play_to_callee;
+		config.allowredirect_in = allowredir_in;
+		config.allowredirect_out = allowredir_out;
+		config.allowdisconnect = allowdisconnect;
+		config.timelimit = timelimit;
+		config.play_warning = play_warning;
+		config.warning_freq = warning_freq;
+		config.warning_sound = warning_sound;
+		config.end_sound = end_sound;
+		config.start_sound = start_sound;
+		res = ast_bridge_call(chan,peer,&config);
 
 		if (res != AST_PBX_NO_HANGUP_PEER)
 			ast_hangup(peer);